Enabling rtcp-rsize negotiation and fixing some issues with it.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 50 matching lines @@
 
 std::string VideoSendStream::Config::Rtp::ToString() const {
   std::stringstream ss;
   ss << "{ssrcs: [";
   for (size_t i = 0; i < ssrcs.size(); ++i) {
     ss << ssrcs[i];
     if (i != ssrcs.size() - 1)
       ss << ", ";
   }
   ss << ']';
+  ss << ", rtcp_mode: "
+     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
+                                          : "RtcpMode::kReducedSize");
   ss << ", max_packet_size: " << max_packet_size;
   ss << ", extensions: [";
   for (size_t i = 0; i < extensions.size(); ++i) {
     ss << extensions[i].ToString();
     if (i != extensions.size() - 1)
       ss << ", ";
   }
   ss << ']';
 
   ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
@@ skipping 543 matching lines @@
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
   vie_encoder_.SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc